In traditional corporate network architectures, each corporate location has separate networks for voice and data traffic. Circuit-switched, Time Division Multiplexing (“TDM”) networks have been used for voice traffic and packet-switched networks have been used for data traffic. Typically, leased telephone lines have been used to interconnect the voice networks and data networks at each corporate location.
The Internet Protocol (“IP”) and many other related protocols are maintained by the Internet Engineering Task Force (“IETF”). The IP is one of the most widely used protocols in packet-switched networks. IP networks are relatively inexpensive to deploy as most computer operating systems come with software that implements the IP. IP networks are also very reliable and have been adapted for many new applications. Consequently, IP networks have been used in many corporate network architectures for data networks. Devices known as routers employ the IP to route data packets among network segments. Typically, in corporate network architectures, leased telephone lines are used to interconnect gateway routers at each corporate location.
Corporate network architectures have similar configurations for interconnecting voice networks at each corporate location. Each corporate location typically has at least one Private Integrated Services Network Exchange (“PINX”) that provides telephony services to users at that location. Leased telephone circuits are used to interconnect gateway PINXs at each corporate location. This arrangement allows telephone users at one corporate location to place telephone calls to users at other corporate locations, without incurring toll charges from public telephone network providers.
Corporations are continually striving to reduce operational costs that are associated with corporate network architectures. A recent trend has been to use corporate packet-switched IP networks to transport both voice and data traffic, so that only one network is required. These networks employ higher level protocols that work with the IP to support telephony applications. Examples of such higher level protocols include the Session Initiation Protocol (“SIP”) and H.323 (which is a standard approved by the International Telecommunication Union (ITU) that defines how audiovisual conferencing data is transmitted across networks by defining protocols to provide audio-visual communication sessions on any packet network.)
Such different architectures implicate many diverse protocols.
So called Q Signaling (“QSIG”) telecommunications standards have been developed in conjunction with the European Computer Manufacturers Association (“ECMA”). A parallel set of standards is maintained by the International Standards Organization (“ISO”). QSIG provides a standard means for establishing, terminating, and clearing voice calls in a Private Integrated Services Network (“PISN”). QSIG has been adopted by most PINX manufacturers to ensure interoperability among PINXs that are made by other manufacturers.
The QSIG protocols allow supplementary services to be provided to users of a PISN. Call Diversion and Single Step Call Transfer are examples of such supplementary services. Call Diversion involves retargeting a call during call establishment by changing a user identity that is used as the basis for routing the call to a destination. Call diversion has several variations, including Call Forwarding Busy (“CFB”), Call Forwarding No Reply (“CFNR”), Call Forwarding Unconditional (“CFU”), and Call Diversion (“CD”). Single Step Call Transfer supplementary services enable a user to transform two of that user's calls, at least one of which must be answered, into a new call between the two other users.
ECMA standards have been developed in support of QSIG supplementary services. A generic functional protocol for supporting supplementary services is defined in Standard ECMA-165. A Call Diversion supplementary service is specified in Standard ECMA-173 and Standard ECMA-174. A Call Transfer supplementary service is defined in Standard ECMA-177 and Standard ECMA-178. A Single Step Call Transfer supplementary service is specified in Standard ECMA-299 and Standard ECMA-300.
Interworking between QSIG and H.323 is defined in Standard ECMA-307. Interworking between QSIG and H.323 for Call Transfer supplementary services is specified in Standard ECMA-308. Interworking between QSIG and H.323 for Call Diversion supplementary services is defined in Standard ECMA-309. Interworking between QSIG and H.323 for Call Completion supplementary services is specified in Standard ECMA-326. Interworking between QSIG and H.323 for basic services is defined in Standard ECMA-332. Tunneling of QSIG through H.323 Networks is specified in Standard ECMA-333. Interworking between QSIG and SIP is defined in Standard ECMA-339. Tunneling of QSIG through SIP networks is specified in Standard ECMA-355.
The SIP is specified in Request For Comments (“RFC”) 3261. An extension to the SIP for providing reliable provisional response messages is defined in RFC 3262. A privacy mechanism for the SIP is defined in RFC 3323. A Refer method for the SIP is specified in RFC 3515. A replaces header for the SIP is defined in RFC 3891. A referred-by mechanism for the SIP is defined in RFC 3892. The H.323 protocol suite was created by the International Telecommunications Union (“ITU”). A call control protocol in H.323 networks is defined in H.225. Security mechanisms for H.323 networks are defined in H.235. A media control protocol is specified in H.245. A generic functional protocol for supporting supplementary services in H.323 networks is defined in H.450.1. While the preceding examples are based on the SIP, it will be understood that the present invention also applies to H.323 networks.
However, despite trends toward integrated architectures as mentioned above, due to substantial investments that previously have been made in circuit-switched voice networks, separate voice and data networks will likely continue to be employed for years to come. Thus, there is a need to allow the coexistence of old infrastructure and new technology—i.e., to allow new IP networks to be connected to heritage QSIG networks.
To this end, gateway devices have been used to facilitate communications among networks that employ different communications protocols. For example, a QSIG/IP Gateway has two interfaces, one for an IP network and a separate interface for a QSIG TDM network. The gateway device performs protocol conversion functions, which allow telecommunications devices that reside in each network to communicate with one another. More particularly, the QSIG/IP Gateway is used to establish, modify, and terminate voice sessions between users of a QSIG PISN and IP network users.
Interworking between a QSIG network and an IP network permits calls originated in the QSIG network to be terminated in the IP network, and calls originating in the IP network to be terminated in the QSIG network. However, prior art gateways that perform QSIG/IP interworking have implemented non-optimal solutions that do not make efficient use of a QSIG/IP gateway's resources. For example, non-optimal QSIG/SIP interworking of the QSIG Call Diversion and Single Step Call Transfer supplementary services has resulted in connections being established through gateway PINXs that are not required. As a result, these prior art gateway PINXs are not utilized efficiently, which reduces the performance of these gateways.